1. Field of the Invention
The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. More specifically, the present invention relates to a transmitter and a receiver using an additional bit allocation method while maintaining bit compatibility so as to improve performance of a conventional speech coder. The transmitter and the receiver according to the present invention may be applicable to a VoIP (Voice-Over Internet Protocol) communication system.
2. Description of the Related Art
Various coding methods have been proposed to convert a voice signal into a digital signal and process the digitalized voice signals. Most popular coding methods may be classified as a waveform coding method such as a PCM (pulse code modulation) method or a hybrid coding method. The hybrid coding method is a combination of a waveform coding method and a parametric coding method. For example, a CELP (code-exited linear prediction) method that is recommended as a standard of ITU-T (International Telecommunication Union-Telecommunication standardization sector) may use the hybrid coding method. Most of the hybrid coding methods are based on a speech production model for effective compression of a voice signal. According to the hybrid coding methods, the voice signal is classified as an excited signal, and spectrum information represents a vocal tract transfer function. The classified spectrum information and the excited signal are respectively modeled and quantized with a predefined method. The quantized spectrum information and the excited signal are transmitted to a receiver. A representative hybrid coding method may be exemplified as an AMR (Adaptive Multi-Rate) coder. The AMR coder is scheduled to be used in the IMT-2000 communication system.
With reference to the G.723.1 standard, it is a standardized algorithm for compressing a multimedia signal by using a minimum number of bits. The G.723.1 algorithm compresses an input voice signal or restores an original uncompressed signal from the input voice signal at two bit rates, such as 5.3 kbit/s and 6.3 kbit/s. The G.723.1 algorithm also provides toll quality equal to the quality level required in a wired network. Similarly, the G.729 algorithm compresses an input voice signal or restores an original uncompressed signal from the input voice signal at a bit rate of 8 kbit/s, and it also provides toll quality equal to the quality level required in a wired network. The G.729 algorithm is widely used in the VoIP application field together with the G.723.1 algorithm. Moreover, the G.729A algorithm is also widely used because it has reduced complexity and has bit compatibility with the G.729 algorithm that requires much computation ability for effective realization. Furthermore, an AMR coder is proposed for the next generation voice communication. There are AMR-NB (AMR-narrowband) coder for processing a telephone band voice signal and AMR-WB (AMR-wideband) for processing a wideband signal.
The above-described voice coders are presently used or scheduled to be used in a wired and wireless voice communication system. The above voice coders quantize spectrum information of voice signals and excited signal information by using a CELP algorithm on the basis of a speech production model. However, there is a problem in that performance deterioration arises in transition frame or with respect to any signal except a voice signal, such as a music signal, since the coders use restricted bit rates. In particular, the G.729 algorithm has a frame size of 10 ms for analyzing parameters, which is less than that of other coders. Accordingly, the G.729 algorithm is appropriate for modeling of the excited signal, but it has a problem in quantization of spectrum information such as LPC. This is because the number of bits to be allocated as linear prediction coefficients (LPC) for quantization in the G.729 algorithm is relatively small.
However, the G.723.1 algorithm has a frame size of 30 ms, which is relatively large. In the case of the G.723.1 algorithm, a sufficient numbers of bits are used for LPC quantization, thus the distortion of the quantized information is reasonable. However, since the G.723.1 uses a linear interpolation method implemented at each interval of the sub-frames, a problem of distortion of spectrum information becomes larger at each sub-frame. In the search duration of a fixed codebook for representing non-periodic excited signals of the coders using the two algorithms, an algebraic codebook comprised of a few pulses is used. Therefore, a problem arises in that the quality is degraded due to a deficiency of the number of pulses for representing the excited signals in any duration, such as the transition duration, whereby performance of an adaptive codebook is degraded.